Previously, the only way to deploy an application-layer control protocol service such as Session Initiation Protocol (SIP) within a stimulus-based network was to have a Call Management Server (CMS) support SIP, and provide interwork SIP line signaling with either Media Gateway Control Protocol (MGCP) trunk signaling or Network-Band Call Signaling (NCS). Enterprise and Service Provider Voice over Internet Protocol (VoIP) deployments are increasingly using SIP, as the session protocol of choice. These sessions may include Internet telephone calls, multimedia distribution, and multimedia conferences.
However, SIP is a text-based protocol, and the parsing and building of messages may put a heavy processing or throughput load on various network elements. As the number of SIP sessions increases, so too the processing load on various devices may also increase. This increase can create a bottleneck in some deployments. Worse, some non-terminal, or intermediately positioned network devices like Session Border Controllers handling SIP-to-SIP calls may actually handle two related instances of SIP sessions, one sending and one receiving, thus leading to a rapid increase in processor and memory utilization. This processing and parsing burden may negatively impact system scalability, and/or reliability. Accordingly, there is a need in the art to provide SIP services that are scalable, reliable, and that reduce the processing load on intermediate network elements.
Embodiments of the present invention and their advantages are best understood by referring to the detailed description that follows. It should be appreciated that like reference numerals are used to identify like elements illustrated in the figures.